/*
  * Copyright (c) 2012 Andrey Utkin
  * Copyright (c) 2012 Stefano Sabatini
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**
  * @file
  * Filter that changes number of samples on single output operation
  */
 
 #include "libavutil/audio_fifo.h"
 #include "libavutil/avassert.h"

 #include "libavutil/channel_layout.h"

 #include "libavutil/opt.h"
 #include "avfilter.h"
 #include "audio.h"

 #include "internal.h"

 #include "formats.h"
 
 typedef struct {
     const AVClass *class;
     int nb_out_samples;  ///< how many samples to output
     AVAudioFifo *fifo;   ///< samples are queued here
     int64_t next_out_pts;
     int pad;
 } ASNSContext;
 
 #define OFFSET(x) offsetof(ASNSContext, x)

 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption asetnsamples_options[] = {

     { "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
     { "n",              "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
     { "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS },
     { "p",   "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS },
     { NULL }

 };
 

 AVFILTER_DEFINE_CLASS(asetnsamples);

 static av_cold int init(AVFilterContext *ctx)

 {
     ASNSContext *asns = ctx->priv;
 
     asns->next_out_pts = AV_NOPTS_VALUE;

     av_log(ctx, AV_LOG_VERBOSE, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad);

 
     return 0;
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     ASNSContext *asns = ctx->priv;
     av_audio_fifo_free(asns->fifo);
 }
 
 static int config_props_output(AVFilterLink *outlink)
 {
     ASNSContext *asns = outlink->src->priv;
 

     asns->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, asns->nb_out_samples);

     if (!asns->fifo)
         return AVERROR(ENOMEM);

     outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;

 
     return 0;
 }
 
 static int push_samples(AVFilterLink *outlink)
 {
     ASNSContext *asns = outlink->src->priv;

     AVFrame *outsamples = NULL;

     int ret, nb_out_samples, nb_pad_samples;

 
     if (asns->pad) {
         nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
         nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
     } else {
         nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
         nb_pad_samples = 0;
     }
 
     if (!nb_out_samples)
         return 0;
 

     outsamples = ff_get_audio_buffer(outlink, nb_out_samples);

     if (!outsamples)
         return AVERROR(ENOMEM);

 
     av_audio_fifo_read(asns->fifo,
                        (void **)outsamples->extended_data, nb_out_samples);
 
     if (nb_pad_samples)
         av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,

                                nb_pad_samples, outlink->channels,

                                outlink->format);

     outsamples->nb_samples     = nb_out_samples;
     outsamples->channel_layout = outlink->channel_layout;
     outsamples->sample_rate    = outlink->sample_rate;

     outsamples->pts = asns->next_out_pts;
 
     if (asns->next_out_pts != AV_NOPTS_VALUE)
         asns->next_out_pts += nb_out_samples;
 

     ret = ff_filter_frame(outlink, outsamples);
     if (ret < 0)
         return ret;

     return nb_out_samples;
 }
 

 static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)

 {
     AVFilterContext *ctx = inlink->dst;
     ASNSContext *asns = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     int ret;

     int nb_samples = insamples->nb_samples;

 
     if (av_audio_fifo_space(asns->fifo) < nb_samples) {
         av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
         ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
         if (ret < 0) {
             av_log(ctx, AV_LOG_ERROR,
                    "Stretching audio fifo failed, discarded %d samples\n", nb_samples);

             return -1;

         }
     }
     av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
     if (asns->next_out_pts == AV_NOPTS_VALUE)
         asns->next_out_pts = insamples->pts;

     av_frame_free(&insamples);

     while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)

         push_samples(outlink);

     return 0;

 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterLink *inlink = outlink->src->inputs[0];
     int ret;
 

     ret = ff_request_frame(inlink);

     if (ret == AVERROR_EOF) {

         ret = push_samples(outlink);
         return ret < 0 ? ret : ret > 0 ? 0 : AVERROR_EOF;

     }

 
     return ret;
 }
 

 static const AVFilterPad asetnsamples_inputs[] = {
     {

         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,

     },

     { NULL }

 };
 
 static const AVFilterPad asetnsamples_outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .request_frame = request_frame,
         .config_props  = config_props_output,
     },
     { NULL }
 };
 

 AVFilter ff_af_asetnsamples = {

     .name        = "asetnsamples",
     .description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
     .priv_size   = sizeof(ASNSContext),
     .priv_class  = &asetnsamples_class,
     .init        = init,
     .uninit      = uninit,
     .inputs      = asetnsamples_inputs,
     .outputs     = asetnsamples_outputs,

 };